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EP3584927B1 - Systèmes et procédés de traitement d'un signal audio pour relecture sur un dispositif audio - Google Patents

Systèmes et procédés de traitement d'un signal audio pour relecture sur un dispositif audio Download PDF

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Publication number
EP3584927B1
EP3584927B1 EP18178873.8A EP18178873A EP3584927B1 EP 3584927 B1 EP3584927 B1 EP 3584927B1 EP 18178873 A EP18178873 A EP 18178873A EP 3584927 B1 EP3584927 B1 EP 3584927B1
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EP
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Prior art keywords
signal
drc
subband signal
feedforward
subband
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EP18178873.8A
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German (de)
English (en)
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EP3584927A1 (fr
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Nicholas R. Clark
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Mimi Hearing Technologies GmbH
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Mimi Hearing Technologies GmbH
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Priority to EP18178873.8A priority Critical patent/EP3584927B1/fr
Priority to US16/019,230 priority patent/US10199047B1/en
Priority to CN201811064528.XA priority patent/CN109545233B/zh
Priority to KR1020180135149A priority patent/KR102343447B1/ko
Priority to US16/244,727 priority patent/US10991375B2/en
Priority to CA3099725A priority patent/CA3099725A1/fr
Priority to PCT/EP2019/051041 priority patent/WO2019242895A1/fr
Priority to AU2019289648A priority patent/AU2019289648B2/en
Priority to JP2020569897A priority patent/JP7237993B2/ja
Priority to EP19152078.2A priority patent/EP3584928B1/fr
Priority to US16/440,169 priority patent/US11062717B2/en
Priority to EP19180587.8A priority patent/EP3595172B1/fr
Publication of EP3584927A1 publication Critical patent/EP3584927A1/fr
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/0204—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders using subband decomposition
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G9/00—Combinations of two or more types of control, e.g. gain control and tone control
    • H03G9/005—Combinations of two or more types of control, e.g. gain control and tone control of digital or coded signals
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G7/00—Volume compression or expansion in amplifiers
    • H03G7/002—Volume compression or expansion in amplifiers in untuned or low-frequency amplifiers, e.g. audio amplifiers
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/26—Pre-filtering or post-filtering
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G7/00—Volume compression or expansion in amplifiers
    • H03G7/007—Volume compression or expansion in amplifiers of digital or coded signals
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G9/00—Combinations of two or more types of control, e.g. gain control and tone control
    • H03G9/02—Combinations of two or more types of control, e.g. gain control and tone control in untuned amplifiers
    • H03G9/025—Combinations of two or more types of control, e.g. gain control and tone control in untuned amplifiers frequency-dependent volume compression or expansion, e.g. multiple-band systems
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G9/00—Combinations of two or more types of control, e.g. gain control and tone control
    • H03G9/20—Combinations of two or more types of control, e.g. gain control and tone control in frequency-selective amplifiers
    • H03G9/24—Combinations of two or more types of control, e.g. gain control and tone control in frequency-selective amplifiers having semiconductor devices
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G9/00—Combinations of two or more types of control, e.g. gain control and tone control
    • H03G9/26—Combinations of two or more types of control, e.g. gain control and tone control in untuned amplifying stages as well as in frequency-selective amplifying stages
    • H03G9/30—Combinations of two or more types of control, e.g. gain control and tone control in untuned amplifying stages as well as in frequency-selective amplifying stages all amplifying stages having semiconductor devices
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/35—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception using translation techniques
    • H04R25/356—Amplitude, e.g. amplitude shift or compression
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2205/00—Details of stereophonic arrangements covered by H04R5/00 but not provided for in any of its subgroups
    • H04R2205/041—Adaptation of stereophonic signal reproduction for the hearing impaired
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/43—Signal processing in hearing aids to enhance the speech intelligibility
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/03—Synergistic effects of band splitting and sub-band processing

Definitions

  • This invention relates generally to the field of digital signal processing (DSP), audio engineering and audiology - more specifically systems and methods for processing an audio signal for replay on an audio device, for example for providing an enhanced listening experience on the audio device.
  • DSP digital signal processing
  • Listeners with sensorineural hearing loss typically have similar, or even reduced, discomfort thresholds when compared to normal hearing listeners, despite their hearing thresholds being raised. To this extent, their dynamic range is narrower and simply adding EQ gain would be detrimental to their hearing health in the long run ( FIG.1 ).
  • Dynamic range compression can be used to address this issue by amplifying quieter sounds while reducing the volume of loud sounds, thus narrowing the dynamic range of the audio.
  • DRC Dynamic range compression
  • hearing aid processors employ wide dynamic range compression where the faintest sounds are amplified considerably, but where high-intensity sounds are not.
  • conventional hearing aids are designed for use in real world situations where a wide dynamic range of sounds are relevant to the listener, i.e.
  • the listener wants to make sense of sonic information such as a loud-voiced person speaking in front of them, while at the same time they want to be able to detect the faint sound of a car approaching them from a distance while walking down the street.
  • sonic information such as a loud-voiced person speaking in front of them
  • audio content consumed on mobile devices, or other similar devices have very different signal statistics to the sounds that someone will encounter in their daily life, so a different processing strategy is required to provide the listener with a beneficial sound personalization experience.
  • the ability to digitally recreate the functional processing of healthy human hearing would enable a more natural and clear listening experience for a hearing impaired (HI) user.
  • HI hearing impaired
  • the human ear pre-processes sounds into a format that is optimal for transmission to the brain to make sense of the sonic environment.
  • the pre-processing can be modelled as a number of hierarchical signal processes and feedback loops, many of which are non linear, resulting in a complex, non-linear system.
  • hearing loss typically begins at higher frequencies, listeners who are aware that they have hearing loss do not typically complain about the absence of high frequency sounds. Instead, they report difficulties listening in a noisy environment and in hearing out the details in a complex mixture of sounds, such as in an audio stream. In essence, off frequency sounds more readily mask information with energy in other frequencies for HI individuals - music that was once clear and rich in detail becomes muddled. This is because music itself is highly self-masking.
  • a raised threshold in an audiogram is not merely a reduction in aural sensitivity, but a result of the malfunction of some deeper processes within the auditory system that have implications beyond the detection of faint sounds.
  • the BioAid algorithm includes decomposition of a received audio signal into bands, wherein each band signal is band pass filtered and modulated based on a delayed feedback attenuation control signal, prior to compression in an instantaneous compressor.
  • the compressed signal is used to form the delayed feedback attenuation control signal and is also band pass filtered to reduce resulting spread of signal energy.
  • this algorithm served merely as a framework for modeling the hearing system and was not specifically designed for sound augmentation. To this extent, it has some drawbacks on the subjective hearing experience caused by the lack of control over the distortion products. These include a reduced ability to control distortion, a limited frequency resolution and phase distortion that can cause temporal smearing of sound (if used in combination with narrowband filters) and therefore reduced clarity. Namely, although this algorithm could potentially improve some aspects of real world use cases if used by hard of hearing users, it would fail to improve the listening experience for a broader category of listeners in the context of audio. Accordingly, it is the object of this invention to create an improved, biologically-inspired DSP that provides a listener with beneficial sound personalization.
  • the presented technology solves the limitations inherent in prior art DSP methodologies, namely poor frequency resolution and temporal smearing caused by group delay differences between bands.
  • the invention provides an enhanced listening experience on an audio device for both hard of hearing listeners as well as individuals with low to moderate hearing loss, who experience clearer listening experience.
  • the technology features methods for processing an audio signal for replay on an audio device.
  • the methods may be methods of processing an audio signal to provide an enhanced hearing experience on (e.g., when replayed on) an audio device.
  • a method of processing an audio signal for replay on an audio device comprises: a) performing a spectral decomposition of the audio signal (501) into a plurality of subband signals using a band pass filter (402, 502); b) for each subband signal, providing the subband signal to a respective modulator (407) and from the modulator output, providing the subband signal to a respective first processing path that includes a first dynamic range compressor, DRC (404); c) for each subband signal, feedforward compressing the subband signal by the respective first DRC (404) to obtain a feedforward-compressed subband signal; d) for each subband signal, providing the feedforward-compressed subband signal to a second processing path that includes a second DRC (406), compressing the feedforward-compressed subband signal by the respective second DRC (406), and providing an output of the second processing path to the respective modulator (407), wherein modulating the subband signal by the respective modulator (407) is performed in
  • the second processing path may be referred to as a feedback path.
  • the second DRC may be referred to as a feedback DRC
  • the feedback DRC may be delayed relative to the feedforward DRC. That is, the output of the second processing path may be deliberately delayed, e.g., by a delay element (such a s a buffer, for example). The delay may be inserted before or after the feedback DRC.
  • the (delayed) feedback compressed frequency band is then provided to the modulator for the respective subband.
  • the output of the second processing path is provided to the respective modulator.
  • the modulator may provide attenuation at the compression input of step c, in dependence on the output of the second processing path.
  • the proposed method has the advantage and technical effect of providing an enhanced listening experience for a user. This is achieved by processing an audio signal using techniques that mimic the functional processing of the healthy human auditory system.
  • the input band pass filter is phase linear.
  • the phase linear input band pass filter is a finite impulse response filter operating in the frequency domain.
  • a method of processing an audio signal as defined in claim 2 is provided to provide an enhanced hearing experience when replayed on an audio device.
  • the input band pass filter is phase linear.
  • the phase linear input band pass filter is a finite impulse response filter operating in the frequency domain.
  • a method of processing an audio signal as defined in claim 3 is provided to provide an enhanced hearing experience on (when replayed on) an audio device.
  • the input band pass filter is phase linear.
  • the phase linear input band pass filter is a finite impulse response filter operating in the frequency domain.
  • a method of processing an audio signal as defined in claim 7 is provided to provide an enhanced hearing experience on (when replayed on) an audio devide.
  • the input band pass filter is phase linear.
  • the phase linear input band pass filter is a finite impulse response filter operating in the frequency domain.
  • DSP methods that mimic the functional processing of the healthy human are vital to the creation of objectively and subjectively enriched listening experiences.
  • audio device is defined as any device that outputs audio, including, but not limited to: mobile phones, computers, televisions, hearing aids, headphones and/or speaker systems.
  • dynamic range compression is defined as an audio process that reduces the dynamic range of an audio signal.
  • a compressor may either have a feedforward or feedback design. Dynamic range compression may occur instantaneously or the rate may be controlled through adjustment of the compressor's attack and release time constants.
  • bandpass filter is defined as a device that (substantially) passes frequencies within a certain range and attenuates frequencies outside that range.
  • phase linear is defined as property wherein the phase response of a filter is a linear function of frequency such that all frequency components of an input signal are shifted in time by the same constant amount, resulting in no phase distortion.
  • harmonic distortion is defined as the generation of multiples of the original frequencies caused by a nonlinear system.
  • intermodulation distortion is defined as the generation of cross-product frequencies produced when two or more signals mix in a nonlinear system.
  • computer readable storage medium is defined as a solid, non-transitory storage medium including, but not limited to: USB storages with flash memory, a CD-ROM, DVD or BluRay®, a disk or a tape. It may also be a physical storage place in a server accessible by a user, e.g. to download for installation of the computer program on her device or for cloud computing.
  • the model of normal hearing consists of cascading stages, simulating the physiological parts of the signal processing pathway in the auditory system.
  • the model developed by Meddis, R. (see Meddis, R., N. R. Clark, W. Lecluyse, and T. Jürgens, "BioAid - A Biologically inspired Hearing Aid”. Audiological Acoustics 52: 2013, 148-152, 2013 ) aimed to provide a faithful representation of auditory nerve firing patterns, as seen in the model of the auditory periphery in FIG. 2 .
  • the acoustic input signal is filtered - mimicking the outer- and middle-ear 201 frequency-dependent transfer of sound pressure to displacement of the stapes.
  • the signal is decomposed into frequency bands using filter bank 202.
  • the filter bank consists of a nonlinear path, modeling the contribution of outer hair cells and a linear path, modeling the passive response of the basilar membrane.
  • Subsequent stages simulate stereo displacement, inner hair cell 203 potential fluctuations, and finally transmitter release into the synaptic cleft 204 located between inner hair cells and the auditory nerve 205.
  • the MOC 207 feedback loop is tonotopically implemented as an attenuation of activity in the nonlinear path of the of the filter bank - the amount of attenuation is controlled by the total spiking activity in the corresponding frequency band on the brainstem 208 level.
  • the MOC attenuation of BM response builds up during steady portions of an acoustic stimulus and decays with a time constant 50ms following the offset of the stimulus.
  • a delay of 10 ms between the onset of a stimulus and the beginning MOC attenuation is used to mimic synaptic latencies.
  • the acoustic reflex 206 is implemented as an attenuation of the stapes response on the total activity of all neurons.
  • This healthy hearing model formed the basis of Meddis et al.'s "BioAid-algorithm", as illustrated in FIG. 3 .
  • the audio signal is provided at a control input 301, which is then spectrally decomposed 302 into a plurality of frequency bands by the input band pass filter 303 (which is an infinite impulse response (IIR) filter that is recursive, and, thus, introduces phase distortion).
  • IIR infinite impulse response
  • Each frequency band is processed in the schematically shown plurality of parallel channels having all the same operators, though different parameters to it.
  • Each respective frequency band is provided at a compression input 304, which is feed forward compressed by an instantaneous dynamic range compressor 305 (with time constant zero).
  • the audio signal is processed by a feedback DRC 307, wherein the feedback DRC 307 is delayed relative to the instantaneous DRC 305. Subsequently, the compressed audio signal is modulated via a modulator 308 to attenuate the audio signal provided to the instantaneous, feedforward DRC.
  • the delayed feedback DRC processing is characterized by two adjustable parameters: a threshold parameter and a strength parameter.
  • the threshold parameter specifies the level of the output from the instantaneous, feedforward DRC, at which the feedback processing starts to work.
  • the strength parameter which governs the amount of attenuation applied when the feedback processing is active, is a scalar that is multiplied by the ratio of the input to the feedback-processing process relative to the feedback processing threshold in dB (thus giving the attenuation values in dB).
  • This compressed audio signal from the compression output 306 is passed through an output IIR band pass filter 309 to control the spectral spread of distortion.
  • This secondary IIR filter 309 has the same pass-band as the input IIR band pass filter 303. Subsequently, the compressed frequency bands are modulated by a gain 310 and, finally, recombined in an operator 311 to form a full wide band audio signal again to be provided at the control output.
  • this algorithm has drawbacks on the subjective hearing experience caused by the use and arrangement of IIR filters 303, 309 and the instantaneous DRC 305.
  • This configuration leads to an unconstrained spread of audible distortion, leading to a negative impact on perceived quality, particulary for users with milder forms of hearing loss (see e.g. FIG 6 , 8 ).
  • the algorithm was designed as a hearing aid algorithm to aid hard of hearing listeners in real world use cases (e.g. speech in noise), and thus would be a poor model for personalizing audio content.
  • FIGS. 4 and 5 an example embodiment of the invention is illustrated.
  • a wide band audio signal is provided at processing input 401 and then spectrally decomposed 501 into a plurality of frequency bands (into a plurality of subband signals, each subband signal in a respective frequency band).
  • Spectral decomposition is performed by the input band pass filter 402, 502.
  • Each respective frequency band is provided at a compression input 403, which is feed forward compressed by feedforward DRC 404.
  • Feedforward DRC 404 (or the dynamic range compression that is applied by the DRC 404) is slowed relative to an instantaneous DRC. This slowing is indirect, which is in accordance with the invention, or may be direct, which is not in accordance with the invention. Indirectly slowing a feedforward compression is achieved through means other than adjusting a DRC's attack and/or release time constants. This occurs through the use of a finite impulse response filter followed by signal oversampling, which has the net effect of slowing down feedforward compression relative to an instantaneous DRC. In this case, the compressed frequency bands are downsampled, e.g.
  • feedforward DRC 404 is slowed directly through adjusting DRC 404's attack and/or release time constants. This slowed compression, in both cases, results in the reduction of the spectral spread of harmonic distortion and intermodulation distortion products relative to the prior art.
  • this arrangement also replaces the need for having output band pass filter 309 - which results in a more symmetrical distribution of the harmonic distortion that remains.
  • natural distortion emanates symmetrically at all frequency regions of the basilar membrane as the cochlear process, in itself, can be thought of as a resonant system with a continuum of changing properties.
  • FIG. 8 when the distortion pattern of a given frequency is moved towards one of the band edges of an output band pass filter, see panels 802, 803, the distortion pattern is asymmetrically curtailed, see panel 802.
  • harmonic symmetry is better retained - thus better emulating the natural distortion creating by the basilar membrane.
  • the audio signal is processed by a feedback DRC 406.
  • the feedback DRC 406 is delayed relative to the feedforward DRC.
  • the feedback pathway is tapped from the output of the feedforward DRC process.
  • the feedback pathway may be attenuated by thresholding to obtain signal parts above a certain threshold.
  • This signal may then be low-pass filtered for temporal smoothing, and may be multiplied by a scalar factor.
  • the aforementioned delay may be achieved through the use of a buffer, such as a ring buffer, for example.
  • This stream of values is subsequently used to modulate the audio signal provided to the feedforward DRC 404 within each band. Modulation, feedforward compression and feedback compression proceed in a continuous manner.
  • the feedback loop dynamically adapts compression to the audio signal level, enabling more effective mitigation of off-frequency sound masking - a process that physiologically occurs in the auditory system.
  • Frequency bands may be modulated by a gain 408 and, finally, recombined in operator 503 to form a full wide audio band signal again to be provided at the control output 504.
  • Each frequency band may have its own, distinct parameters, e.g. gain, attenuation factors, etc.
  • FIG. 9 Another example embodiment of the invention is illustrated in FIG. 9 , in which sound processed according the configuration of FIGS. 4 , 5 is split into a first and second signaling pathway (or signal pathway).
  • a wide band audio signal is provided at processing input 901 and then divided into a first pathway (first signal pathway) 902 and a second pathway (second signal pathway) 903.
  • the second pathway 903 is only subject to a delay 904 and a protective limiter 905.
  • the audio signal from the control input 901 is spectrally decomposed and processed according to the configuration of FIG. 4 .
  • Each pathway 902, 903 may include a weighting operator 906 and 907, respectively.
  • these weighting operators 906 and 907 may be correlated by a common function that may be adjustable by a user by one single control variable 910. Then these pathways 902 and 903 are recombined according to their weighting factors in operator 908 and provided to the processing output 909.
  • Parallel compression provides the benefit of allowing the user to mix 'dry' unprocessed or slightly processed sound with 'wet' processed sound, enabling customization of processing based on subjective preference. For example, this enables hearing impaired users to use a high ratio of heavily processed sound relative to users with moderate to low hearing loss. Furthermore, by reducing the dynamic range of an audio signal by bringing up the softest sounds, rather than reducing the highest peaks, it provides audible detail to sound. The human ear is sensitive to loud sounds being suddenly reduced in volume, but less sensitive to soft sounds being increased in volume, and this mixing method takes advantage of this observation, resulting in a more natural sounding reduction in dynamic range compared with using a dynamic range compressor in isolation. Additionally, parallel compression is in particular useful for speech-comprehension and/or for listening to music with full, original timbre.
  • the pathway mixing modulator involves a phase correction network in order to prevent any phase cancellations upon summing the correlated signals to provide an audio signal to the control output.
  • parallel compression is problematic using the approach in the prior art as the recursive input and output IIR band pass filters introduce phase distortion into the audio signal.
  • superposition of a phase-distorted signal with a similar audio signal can cause so-called comb-filtering effects, which adversely affects the timbral quality of the results. Users are sensitive to these effects, which are detrimental to the subjective hearing experience.
  • FIG. 10 A further example is illustrated in FIG. 10 , in which a wide band audio signal is provided at processing input 1001 and then divided into a first processed pathway 1002 (first signaling pathway, or first signal pathway), utilizing the same processing configuration as FIG. 3 , and a second processed pathway 1003 (first signaling pathway, or first signal pathway).
  • first processed pathway 1002 first signaling pathway, or first signal pathway
  • second processed pathway 1003 first signaling pathway, or first signal pathway
  • the second processed pathway 1003 may include, for instance, a sound enhancing algorithm, such as a speech-comprehension algorithm, to allow the user to adjust pathway ratios to allow for better speech comprehension and/or to have a subjectively more comfortable music hearing experience for the user in a respective background noise environment.
  • a sound enhancing algorithm such as a speech-comprehension algorithm
  • any sound processing algorithm such as a hearing aid algorithm, preferably having linear phase characteristics, could be mixed with the first processed pathway 1002.
  • the two processed pathways have to be in line with one of the following rules.
  • the signals in each pathway must either (i) conform to phase linearity, (ii) introduce identical phase distortions, or (iii) involve a phase correction network in order to prevent any phase cancellations upon summing the correlated signals to form the output.
  • option (i) or option (ii) is implemented as this allows for lean and simple implementation.
  • FIGS. 11 and 12 A further example embodiment of the invention is illustrated in FIGS. 11 and 12 , in which a wide band audio signal is provided at processing input 1101, 1201 and then spectrally decomposed into a plurality of frequency bands (into a plurality of subband signals, each subband signal in a respective frequency band). Spectral decomposition is performed by the input band pass filter 1202. Each respective frequency band (e.g., subband signal) is divided into a first pathway (first signal pathway) 1103 and a second pathway (second signal pathway) 1104. In this example, the second pathway 1104 is lightly processed as it only includes a delay 1105 and a protective limiter 1106. In contrast, first pathway 1103 is processed similarly to the configuration illustrated in FIG. 5 .
  • first pathway 1103 is processed similarly to the configuration illustrated in FIG. 5 .
  • the first pathway frequency band is provided at a compression input, which is feed forward compressed by feedforward DRC 404.
  • the audio signal is processed by a feedback DRC 406, wherein the feedback DRC 406 may be delayed relative to the feedforward DRC 404. That is, the output of the feedback processing pathway may be deliberately delayed, e.g. by a delay element (such as a buffer, for example). The delay may be inserted before or after the feedback DRC.
  • the compressed audio signal may be modulated via a modulator 407 to attenuate the audio signal provided to feedforward DRC 404.
  • the compressed frequency bands may then be modulated by a gain 408, 1107.
  • First and second frequency band pathways may include a weighting operator 1108 and 1109, respectively.
  • these weighting operators 1108 and 1109 may be correlated by a common function that may be adjustable by a user by one single control variable 1112.
  • these pathways 1103 and 1104 are recombined according to their weighting factors in operator 1110 and provided to the processing output 1111.
  • the frequency bands e.g., subband signals
  • FIGS. 11 and 12 importantly allows the user much more control over which frequencies are processed in the audio spectrum of a signal. For instance, in a music composition with intense treble, it may be preferable to process sounds across a higher frequency range. Conversely, while focusing on human speech, approximately around 150Hz-4kHz, processing may focus on narrower spectra. Generally, spectral processing can be adjusted for computational savings purposes.
  • FIG. 13 A further example embodiment of the invention is illustrated in FIG. 13 .
  • a wide band audio signal is provided at processing input 1301 and then spectrally decomposed into a plurality of frequency bands (e.g., into a plurality of subband signals, each subband signal in a respective frequency band).
  • Spectral decomposition is performed by the input band pass filter 1302.
  • Each respective frequency band is provided at a compression input 1303, which is feed forward compressed by feedforward DRC 1304.
  • Feedforward DRC 1304 is slowed relative to an instantaneous DRC. This occurs indirectly, in the manner described above.
  • Fig. 13 shows the feeding of weighted feedforward compressed audio signals from neighboring frequency bands for a single frequency band only, it is understood that feedback compression in each frequency band receives weighted feedforward compressed audio signals as inputs from respective neighboring frequency bands.
  • the feedback DRC 1314 may be delayed relative to the feedforward DRC 1304. That is, the output of the feedback DRC may be deliberately delayed, e.g., by a delay element (such as a buffer for example). The delay may be inserted before or after the feedback DRC.
  • the plurality of delayed feed-back compressed audio signals may be modulated via a modulator 1310 to attenuate the audio signal provided to feedforward DRC 1304.
  • the compressed frequency bands may then be modulated by a gain 1311 and, finally, recombined in operator 1312 to form a full wide audio band signal again to be provided at the control output 1313.
  • the ability to attenuate the audio signal in a given frequency band as a function that includes signal levels in one or more neighboring frequency bands provides a more refined degree of parameterization for an augmented hearing experience.
  • this process happens naturally in the olivocochlear system in the ear ( Effects of electrical stimulation of efferent olivochoclear neurons on cat auditory-nerve fibers. III, Tuning curves and thresholds at CF, Guinan & Gifford, 1988 ) - and thus this audio processing configuration more closely models healthy hearing in the auditory stem.
  • the presented technology creates improved, biologically-inspired DSP algorithms that more closely mimic the functional processing of the healthy human ear.
  • the invention avoids the limitations inherent in prior art DSP methodologies, namely unconstrained frequency distortion and phase distortion. To this extent, the invention provides an enhanced listening experience both to hard of hearing individuals as well as individuals with healthy hearing, who experience a richer, crisper listening experience of audio content.

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  • Signal Processing (AREA)
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  • Audiology, Speech & Language Pathology (AREA)
  • Computational Linguistics (AREA)
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  • Otolaryngology (AREA)
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Claims (14)

  1. Procédé de traitement d'un signal audio pour relecture sur un dispositif audio, ce procédé comprenant :
    a. la réalisation d'une décomposition spectrale du signal audio (501) en une pluralité de signaux de sous-bande en utilisant un filtre passe-bande (402, 502) ;
    b. pour chaque signal de sous-bande, la fourniture du signal de sous-bande à un modulateur zone blanche respectif (407) et, depuis la sortie de modulateur, la fourniture du signal de sous-bande à un premier parcours de traitement respectif qui inclut un premier compresseur à plage dynamique DRC (404) ;
    c. pour chaque signal de sous-bande, la compression prédictive du signal de sous-bande par le premier à plage dynamique DRC respectif (404) pour obtenir un signal de sous-bande à compression prédictive ;
    d. pour chaque signal de sous-bande, la fourniture du signal de sous-bande à compression prédictive à un second parcours de traitement qui inclut un second DRC (406), la compression du signal de sous-bande à compression prédictive par le second DRC respectif (406), et la fourniture d'une sortie du second parcours de traitement au modulateur respectif (407), la modulation du signal de sous-bande par le modulateur respectif (407) étant réalisée en fonction de la sortie du second parcours de traitement ; et
    e. la recombinaison des signaux de sous-bande à compression prédictive,
    caractérisé en ce que la compression prédictive comprend, pour chaque signal de sous-bande, le ralentissement du premier DRC respectif (404) par rapport à un DRC instantané utilisant un filtre à réponse d'impulsion finie, suivi par un suréchantillonnage de signal.
  2. Procédé de traitement d'un signal audio pour relecture sur un dispositif audio, ce procédé comprenant la division d'un signal audio non traité en un premier chemin de signal (903, 1003) et un second chemin de signal (902, 1002), le traitement du signal audio dans le premier chemin de signal (902, 1002), et la recombinaison de sorties des premier et second chemins de signal (902, 903 ; 1002, 1003) à un rapport (910, 1004), le traitement dans le premier chemin de signal (902, 1002) comprenant :
    a. la réalisation d'une décomposition spectrale du signal audio (501) en une pluralité de signaux de sous-bande en utilisant un filtre passe-bande (402, 502) ;
    b. pour chaque signal de sous-bande, la fourniture du signal de sous-bande modulateur respectif (407) et, depuis la sortie de modulateur, la fourniture du signal de sous-bande à un premier parcours de traitement respectif qui inclut un premier compresseur à plage dynamique DRC (404) ;
    c. pour chaque signal de sous-bande, la compression prédictive du signal de sous-bande par le premier compresseur à plage dynamique DRC (404) pour obtenir un signal de sous-bande à compression prédictive ;
    d. pour chaque signal de sous-bande, la fourniture du signal de sous-bande à compression prédictive à un second parcours de traitement qui inclut un second DRC (406), la compression du signal de sous-bande à compression prédictive par le second DRC respectif (406), et la fourniture d'une sortie du second parcours de traitement au modulateur respectif (407), la modulation du signal de sous-bande par le modulateur respectif (407) étant réalisée en fonction de la sortie du second parcours de traitement ; et
    e. la recombinaison des signaux de sous-bande à compression prédictive,
    la compression prédictive comprenant, pour chaque signal de sous-bande, le ralentissement du premier DRC respectif (404) par rapport à un DRC instantané utilisant un filtre à réponse d'impulsion finie, suivi par un suréchantillonnage de signal.
  3. Procédé de traitement d'un signal audio pour relecture sur un dispositif audio, ce procédé comprenant :
    a. la réalisation d'une décomposition spectrale (1202) du signal audio en une pluralité de signaux de sous-bande en utilisant un filtre passe-bande (1102, 1202) ;
    b. pour chaque signal de sous-bande, la division du premier signal de sous-bande en un premier chemin de signal (1103) et un second chemin de signal (1104), le traitement du signal de sous-bande dans le premier chemin de signal (1103), et la recombinaison des premier et second chemin de signal (1103, 1104) à un rapport pour obtenir un signal de sous-bande traité ;
    c. la recombinaison des signaux de sous-bande traités, le traitement du signal de sous-bande dans le premier chemin de signal (1103) comprenant
    b1. la fourniture du signal à un modulateur respectif (407), depuis la sortie de modulateur, la fourniture du signal de sous-bande à un premier parcours de traitement respectif qui inclut un premier compresseur à plage dynamique DRC (404) ;
    b2. la compression prédictive du signal de sous-bande par le premier DRC respectif (404) pour obtenir un signal de sous-bande à compression prédictive ; et
    b3. la fourniture du signal de sous-bande à compression prédictive à un second parcours de traitement qui inclut un second DRC (406), la compression du signal de sous-bande à compression prédictive par le second DRC respectif (406), et la fourniture d'une sortie du second parcours de traitement au modulateur respectif (407), la modulation du signal de sous-bande étant réalisée en fonction de la sortie du second parcours de traitement ; et
    la compression prédictive comprenant, pour chaque signal de sous-bande, le ralentissement du premier DRC respectif (404) par rapport à un DRC instantané utilisant un filtre à réponse à impulsion finie, suivi par un suréchantillonnage de signal.
  4. Procédé selon la revendication 2 à 3, dans lequel le rapport (910) est un rapport défini par l'utilisateur.
  5. Procédé selon l'une quelconque des revendications 2 à 4, dans lequel le second chemin de signal (903 ; 1104) caractérise un retard et le signal retardé est soumis à un limiteur protecteur.
  6. Procédé selon l'une quelconque des revendications 2 à 5, dans lequel seules les fréquences entre 125 Hz et 12 000 Hz sont traitées dans le premier chemin de signal (902 ; 1103).
  7. Procédé de traitement d'un signal audio pour relecture sur un dispositif audio, ce procédé comprenant :
    a. la réalisation d'une décomposition spectrale (1302) du signal audio en une pluralité de signaux de sous-bande en utilisant un filtre passe-bande ;
    b. pour chaque signal de sous-bande, la fourniture du signal de sous-bande à un modulateur respectif (407) et, depuis la sortie de modulateur, la fourniture du signal de sous-bande à un premier parcours de traitement respectif qui inclut un premier compresseur à plage dynamique DRC (1304) ;
    c. pour chaque signal de sous-bande, la compression prédictive du signal de sous-bande par le premier compresseur à plage dynamique DRC (1304) pour obtenir un signal de sous-bande à compression prédictive ;
    d. pour chaque signal de sous-bande, la fourniture du signal de sous-bande à compression prédictive à un second parcours de traitement qui inclut un second DRC (1314) et en outre la fourniture d'un ou plusieurs signaux de sous-bande à compression prédictive depuis des bandes de fréquence voisines, chacun pondéré avec un facteur de pondération respectif, au second parcours de traitement, la compression, dans le second parcours de traitement, du signal de sous-bande à compression prédictive et des signaux de sous-bande pondérés à compression prédictive des sous-bandes de fréquence respectives par le second DRC respectif (1314), et la fourniture d'une sortie du second parcours de traitement au modulateur respectif, la modulation du signal de sous-bande étant réalisée en fonction de la sortie du second parcours de traitement ; et
    e. la recombinaison des signaux audio à compression prédictive,
    la compression prédictive comprenant, pour chaque signal de sous-bande, le ralentissement du premier DRC respectif (404) par rapport à un DRC instantané utilisant un filtre à réponse à impulsion finie, suivi par un suréchantillonnage de signal.
  8. Procédé selon l'une quelconque des revendications précédentes, comprenant en outre, pour chaque signal de sous-bande, le retardement de la sortie du second parcours de traitement respectif.
  9. Procédé selon l'une quelconque des revendications précédentes, dans lequel, pour chaque signal de sous-bande, la sortie du second parcours de traitement respectif est retardée à raison d'un retardement qui se situe dans l'intervalle de 5 ms à 20 ms.
  10. Procédé selon l'une quelconque des revendications précédentes, dans lequel le filtre passe-bande (402, 502) est de phase linéaire.
  11. Procédé selon l'une quelconque des revendications précédentes, dans lequel le filtre passe-bande (402, 502) est un filtre à réaction à impulsion finie fonctionnant dans le domaine de fréquence.
  12. Procédé selon l'une quelconque des revendications précédentes, dans lequel le dispositif audio est un dispositif parmi : un téléphone mobile, une tablette, un ordinateur, une télévision, une paire d'écouteurs, une aide auditive ou un système vocal.
  13. Dispositif de sortie audio comprenant :
    un processeur apte à réaliser le procédé selon l'une quelconque des revendications précédentes.
  14. Support de stockage lisible par ordinateur comprenant des instructions qui, une fois exécutées par un processeur d'un dispositif de sortie audio, amènent le processeur à réaliser le procédé selon l'une quelconque des revendications 1 à 12.
EP18178873.8A 2018-06-20 2018-06-20 Systèmes et procédés de traitement d'un signal audio pour relecture sur un dispositif audio Active EP3584927B1 (fr)

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EP18178873.8A EP3584927B1 (fr) 2018-06-20 2018-06-20 Systèmes et procédés de traitement d'un signal audio pour relecture sur un dispositif audio
US16/019,230 US10199047B1 (en) 2018-06-20 2018-06-26 Systems and methods for processing an audio signal for replay on an audio device
CN201811064528.XA CN109545233B (zh) 2018-06-20 2018-09-12 用于处理音频信号以在音频装置上重放的系统和方法
KR1020180135149A KR102343447B1 (ko) 2018-06-20 2018-11-06 오디오 디바이스상에서 재생을 위한 오디오 신호 처리를 위한 시스템들 및 방법들
US16/244,727 US10991375B2 (en) 2018-06-20 2019-01-10 Systems and methods for processing an audio signal for replay on an audio device
PCT/EP2019/051041 WO2019242895A1 (fr) 2018-06-20 2019-01-16 Systèmes et procédés de traitement d'un signal audio pour une relecture sur un dispositif audio
CA3099725A CA3099725A1 (fr) 2018-06-20 2019-01-16 Systemes et procedes de traitement d'un signal audio pour une relecture sur un dispositif audio
AU2019289648A AU2019289648B2 (en) 2018-06-20 2019-01-16 Systems and methods for processing an audio signal for replay on an audio device
JP2020569897A JP7237993B2 (ja) 2018-06-20 2019-01-16 オーディオデバイスで再生するためのオーディオ信号を処理するシステム及び方法
EP19152078.2A EP3584928B1 (fr) 2018-06-20 2019-01-16 Systèmes et procédés de traitement d'un signal audio pour relecture sur un dispositif audio
US16/440,169 US11062717B2 (en) 2018-06-20 2019-06-13 Systems and methods for processing an audio signal for replay on an audio device
EP19180587.8A EP3595172B1 (fr) 2018-06-20 2019-06-17 Systèmes et procédés de traitement d'un signal audio pour relecture sur un dispositif audio

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